Call destruction problems * on Solaris 10 SPARC

Every inbound call I make from a PSTN phone to a SIP number living on my * box ends like this:

Scheduling destruction of call ‘000048b50000224400003977000047d1@’ in 15000 ms
== Spawn extension (cp, 0, 1) exited non-zero on ‘SIP/000060002077777-3baa’
– Executing Hangup(“SIP/000060002077777-3baa”, “”) in new stack
== Spawn extension (cp, h, 1) exited non-zero on ‘SIP/000060002077777-3baa’

Before I added explicit hangups in my dialplan, I saw

== Spawn extension (cp, 000060002077777, 1) exited non-zero on ‘SIP/000060002077777-3243’

but I didn’t think I’d need to expicitly tell asterisk to hang up when one side or the other ends the call? As it stands, I have to do a “sip reload” and re-register before another incoming call will come through before asterisk re-registers itself automatically…I’m fairly new to asterisk, am I missing something? I’ve been perusing the docs and other people’s examples, but most people don’t seem to be setting explicit hangups…any advice would be gratefully received.


maybe i can re-word my question…

when i see “Spawn extension (foo, bar, 1) exited non-zero…” in the console, is this not-normal? in my experience with coding, which is by no means prolific, non-zero exists uuuuusually mean something’s not right. is something not right with my asterisk server when any inside->outside or outside->inside calls ends like this?